similar to: Getting source ip adress of incoming INVITE

I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

Hi all I'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. It is clear that the sip logins have been passed to various persons (probably posted on a forum somewhere inviting to do 'free calls'). Right after the affected password was changed, the message log shows which

Hello list, is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110316/bda02b4d/attachment.htm>

Thanks Scott. I?m taking over for someone else?s code, so I must admit I?m still learning the Agent and Queue concepts. Local channels are something I have not used either. Would local channels essentially be an internal bridge? How would I ?Register Local/number at agent in the queue on behalf of the agent (replace number with the agent's extension number)? From: asterisk-users-bounces

Hi * Users, What is the way from the dial-plan to get the IP address of an incoming or outgoing SIP call? I can see the IP address of the SIP call using 'sip show peers' from the CLI. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

After a migration to asterisk 1.6, I don't receive sip incoming calls anymore. As fas as I understand the SIP debug traces, my server receives the request and reject it: ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ <--- SIP read from UDP:212.27.52.5:5060 ---> INVITE sip:s at 192.168.4.2:5060;transport=udp SIP/2.0 Call-ID: 25151-WW-0eaf098b-2f615ac60 at

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to

Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP headers, then Dial(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

Hi, My dialplan looks like this: [from-external] Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) Exten => _X.,n,Ringing Exten => _X.,n,WaitExten(15) Exten => _X.,n,Congestion() Exten => _X.,n,Hangup() include => test [test] Exten => 8282,1,Noop(--- WE GOT TO

I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent. The SIP header I added, I need to have appear in the INVITE sent to the Agent. It works in chan_sip. I send the call to a macro which does... n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE}) n,Queue(${ARG2}) In PJSIP , this doesn't seem to work. Is

I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_username at sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username [my_provider] type=aor contact=sip:sip.example.com:5060 [my_provider] type=endpoint context=from-my_provider

I've setup Asterisk to work with Broadvoice for both incoming and outgoing calls. I can make outgoing calls, but when I try to receive an incoming call I see the following message on the console: [date] NOTICE[8661]: chan_sip.c:13178 handle_request_invite: Unable to create/find SIP channel for this INVITE It's registered with Broadvoice: Name/username Host Dyn

Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/number at domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: number at domain2.com@domain1.com I need: number at domain2.com I can't use "SIP uri dial", i need

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via:

On Tue, Jun 26, 2018 at 7:06 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 06:57 PM, Dovid Bender wrote: > >> Hi, >> >> My dialplan looks like this: >> [from-external] >> >> Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) >> Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) >> Exten => _X.,n,Noop(CALLED

I am getting this message: Failed to authenticate device <sip:2010 at X.X.X.X>;tag=149853321 for INVITE, code = -1 but it does not report the "connecting" address. Who is failing connecting ? I either need to block someone or fix something - I'm thinking block - but I dont know who. How do I found out the connecting IP? Jerry -------------- next part -------------- An HTML